Information About the Cisco Phone Proxy

The Cisco Phone Proxy on the ASA bridges IP telephony between the corporate IP telephony network and the Internet in a secure manner by forcing data from remote phones on an untrusted network to be encrypted.

Phone Proxy Functionality

Telecommuters can connect their IP phones to the corporate IP telephony network over the Internet securely via the phone proxy without the need to connect over a VPN tunnel as illustrated by Figure 3-1.

Figure 3-1 Phone Proxy Secure Deployment

The phone proxy supports a Cisco UCM cluster in mixed mode or nonsecure mode. Regardless of the cluster mode, the remote phones that are capable of encryption are always forced to be in encrypted mode. TLS (signaling) and SRTP (media) are always terminated on the ASA. The ASA can also perform NAT, open pinholes for the media, and apply inspection policies for the SCCP and SIP protocols. In a nonsecure cluster mode or a mixed mode where the phones are configured as nonsecure, the phone proxy behaves in the following ways:

The TLS connections from the phones are terminated on the ASA and a TCP connection is initiated to the Cisco UCM.

SRTP sent from external IP phones to the internal network IP phone via the ASA is converted to RTP.

In a mixed mode cluster where the internal IP phones are configured as authenticated, the TLS connection is not converted to TCP to the Cisco UCM but the SRTP is converted to RTP.

In a mixed mode cluster where the internal IP phone is configured as encrypted, the TLS connection remains a TLS connection to the Cisco UCM and the SRTP from the remote phone remains SRTP to the internal IP phone.

Since the main purpose of the phone proxy is to make the phone behave securely while making calls to a nonsecure cluster, the phone proxy performs the following major functions:

Creates the certificate trust list (CTL) file, which is used to perform certificate based authentication with remote phones.

Modifies the IP phone configuration file when it is requested via TFTP, changes security fields from nonsecure to secure, and signs all files sent to the phone. These modifications secure remote phones by forcing the phones to perform encrypted signaling and media.

Terminates TLS signaling from the phone and initiates TCP or TLS to Cisco UCM

Inserts itself into the media path by modifying the Skinny and SIP signaling messages.

Terminates SRTP and initiates RTP/SRTP to the called party.

NoteAs an alternative to authenticating remote IP phones through the TLS handshake, you can configure authentication via LSC provisioning. With LSC provisioning you create a password for each remote IP phone user and each user enters the password on the remote IP phones to retrieve the LSC. As an alternative to authenticating remote IP phones through the TLS handshake, you can configure authentication via LSC provisioning. With LSC provisioning you create a password for each remote IP phone user and each user enters the password on the remote IP phones to retrieve the LSC.

Because using LSC provisioning to authenticate remote IP phones requires the IP phones first register in nonsecure mode, Cisco recommends LSC provisioning be done inside the corporate network before giving the IP phones to end-users. Otherwise, having the IP phones register in nonsecure mode requires the Administrator to open the nonsecure signaling port for SIP and SCCP on the ASA.

Supported Cisco UCM and IP Phones for the Phone Proxy

The following release of the Cisco Unified Communications Manager are supported with the phone proxy:

Cisco Unified CallManager Version 4. x

Cisco Unified CallManager Version 5.0

Cisco Unified CallManager Version 5.1

Cisco Unified Communications Manager 6.1

Cisco Unified Communications Manager 7.0

Cisco Unified Communications Manager 8.0

Cisco Unified Communications Manager 8.6

Cisco Unified IP Phones

The phone proxy supports these IP phone features:

Enterprise features like conference calls on remote phones connected through the phone proxy

XML services

The following IP phones in the Cisco Unified IP Phones 7900 Series are supported with the phone proxy:

Cisco Unified IP Phone 7975

Cisco Unified IP Phone 7971

Cisco Unified IP Phone 7970

Cisco Unified IP Phone 7965

Cisco Unified IP Phone 7962

Cisco Unified IP Phone 7961

Cisco Unified IP Phone 7961G-GE

Cisco Unified IP Phone 7960 (SCCP protocol support only)

Cisco Unified IP Phone 7945

Cisco Unified IP Phone 7942

Cisco Unified IP Phone 7941

Cisco Unified IP Phone 7941G-GE

Cisco Unified IP Phone 7940 (SCCP protocol support only)

Cisco Unified Wireless IP Phone 7921

Cisco Unified Wireless IP Phone 7925

Note To support Cisco Unified Wireless IP Phone 7925, you must also configure MIC or LSC on the IP phone so that it properly works with the phone proxy.

CIPC for softphones ( CIPC versions with Authenticated mode only)

Note The Cisco IP Communicator is supported with the phone proxy VLAN Traversal in authenticated TLS mode. We do not recommend it for remote access because SRTP/TLS is not supported currently on the Cisco IP Communicator.

Media Termination Instance Prerequisites

The ASA must have a media termination instance that meets the following criteria:

You must configure one media termination for each phone proxy on the ASA. Multiple media termination instances on the ASA are not supported.

For the media termination instance, you can configure a global media-termination address for all interfaces or configure a media-termination address for different interfaces. However, you cannot use a global media-termination address and media-termination addresses configured for each interface at the same time.

If you configure a media termination address for multiple interfaces, you must configure an address on each interface that the ASA uses when communicating with IP phones.

For example, if you had three interfaces on the ASA (one internal interface and two external interfaces) and only one of the external interfaces were used to communicate with IP phones, you would configure two media termination addresses: one on the internal interface and one on the external interface that communicated with the IP phones.

Only one media-termination address can be configured per interface.

The IP addresses are publicly routable addresses that are unused IP addresses within the address range on that interface.

The IP address on an interface cannot be the same address as that interface on the ASA.

The IP addresses cannot be the same as the Cisco UCM or TFTP server IP address.

For IP phones behind a router or gateway, you must also meet this prerequisite. On the router or gateway, add routes to the media termination address on the ASA interface that the IP phones communicate with so that the phone can reach the media termination address.

Certificates from the Cisco UCM

Import the following certificates which are stored on the Cisco UCM. These certificates are required by the ASA for the phone proxy.

Cisco_Manufacturing_CA

CAP-RTP-001

CAP-RTP-002

CAPF certificate (Optional)

If LSC provisioning is required or you have LSC enabled IP phones, you must import the CAPF certificate from the Cisco UCM. If the Cisco UCM has more than one CAPF certificate, you must import all of them to the ASA.

DNS Lookup Prerequisites

If you have an fully qualified domain name (FQDN) configured for the Cisco UCM rather than an IP address, you must configure and enable DNS lookup on the ASA. For information about the dns domain-lookup command and how to use it to configure DNS lookup, see command reference.

After configuring the DNS lookup, make sure that the ASA can ping the Cisco UCM with the configured FQDN.

You must configure DNS lookup when you have a CAPF service enabled and the Cisco UCM is not running on the Publisher but the Publisher is configured with a FQDN instead of an IP address.

Cisco Unified Communications Manager Prerequisites

The TFTP server must reside on the same interface as the Cisco UCM.

The Cisco UCM can be on a private network on the inside but you need to have a static mapping for the Cisco UCM on the ASA to a public routable address.

If NAT is required for Cisco UCM, it must be configured on the ASA, not on the existing firewall.

ACL Rules

If the phone proxy is deployed behind an existing firewall, access-list rules to permit signaling, TFTP requests, and media traffic to the phone proxy must be configured.

If NAT is configured for the TFTP server or Cisco UCMs, the translated “global” address must be used in the ACLs.

Table 3-1 lists the ports that are required to be configured on the existing firewall:

Table 3-1 Port Configuration Requirements

Address

Port

Protocol

Description

Media Termination

1024-65535

UDP

Allow incoming SRTP

TFTP Server

69

UDP

Allow incoming TFTP

Cisco UCM

2443

TCP

Allow incoming secure SCCP

Cisco UCM

5061

TCP

Allow incoming secure SIP

CAPF Service (on Cisco UCM)

3804

TCP

Allow CAPF service for LSC provisioning

Note All these ports are configurable on the Cisco UCM, except for TFTP. These are the default values and should be modified if they are modified on the Cisco UCM. For example, 3804 is the default port for the CAPF Service. This default value should be modified if it is modified on the Cisco UCM.

NAT and PAT Prerequisites

NAT Prerequisites

If NAT is configured for the TFTP server, the NAT configuration must be configured prior to configuring the TFTP Server (the tftp-server command) under the phone proxy.

If NAT is configured for the TFTP server or Cisco UCMs, the translated “global” address must be used in the ACLs.

PAT Prerequisites

When the Skinny inspection global port is configured to use a non-default port, then you must configure the nonsecure port as the global_sccp_port+443 .

Therefore, if global_sccp_port is 7000, then the global secure SCCP port is 7443. Reconfiguring the port might be necessary when the phone proxy deployment has more than one Cisco UCM and they must share the interface IP address or a global IP address.

/* use the default ports for the first CUCM */

object network obj-10.0.0.1-01

host 10.0.0.1

nat (inside,outside) static interface service tcp 2000 2000

object network obj-10.0.0.1-02

host 10.0.0.1

nat (inside,outside) static interface service tcp 2443 2443

/* use non-default ports for the 2nd CUCM */

object network obj-10.0.0.2-01

host 10.0.0.2

nat (inside,outside) static interface service tcp 2000 7000

object network obj-10.0.0.2-02

host 10.0.0.2

nat (inside,outside) static interface service tcp 2443 7443

Note Both PAT configurations—for the nonsecure and secure ports—must be configured.

When the IP phones must contact the CAPF on the Cisco UCM and the Cisco UCM is configured with static PAT (LCS provisioning is required), you must configure static PAT for the default CAPF port 3804.

Prerequisites for IP Phones on Multiple Interfaces

When IP phones reside on multiple interfaces, the phone proxy configuration must have the correct IP address set for the Cisco UCM in the CTL file.

See the following example topology for information about how to correctly set the IP address:

phones --- (dmz)-----|

|----- ASA PP --- (outside Internet) --- phones

phones --- (inside)--|

In this example topology, the following IP address are set:

Cisco UCM on the inside interface is set to 10.0.0.5

The DMZ network is 192.168.1.0/24

The inside network is 10.0.0.0/24

The Cisco UCM is mapped with different global IP addresses from DMZ > outside and inside interfaces > outside interface.

In the CTL file, the Cisco UCM must have two entries because of the two different IP addresses. For example, if the static statements for the Cisco UCM are as follows:

object network obj-10.0.0.5-01

host 10.0.0.5

nat (inside,outside) static 209.165.202.129

object network obj-10.0.0.5-02

host 10.0.0.5

nat (inside,dmz) static 198.168.1.2

There must be two CTL file record entries for the Cisco UCM:

record-entry cucm trustpoint cucm_in_to_out address 209.165.202.129

record-entry cucm trustpoint cucm_in_to_dmz address 192.168.1.2

7960 and 7940 IP Phones Support

An LSC must be installed on these IP phones because they do not come pre installed with a MIC. Install the LSC on each phone before using them with the phone proxy to avoid opening the nonsecure SCCP port for the IP phones to register in nonsecure mode with the Cisco UCM.

See the following document for the steps to install an LSC on IP phones:

Create an ACL to allow CIPC to register with the Cisco UCM in nonsecure mode.

Configure null-sha1 as one of the SSL encryption ciphers.

Current versions of Cisco IP Communicator (CIPC) support authenticated mode and perform TLS signaling but not voice encryption. Therefore, you must include the following command when configuring the phone proxy instance:

cipc security-mode authenticated

Because CIPC requires an LSC to perform the TLS handshake, CIPC needs to register with the Cisco UCM in nonsecure mode using cleartext signaling. To allow the CIPC to register, create an ACL that allows the CIPC to connect to the Cisco UCM on the nonsecure SIP/SCCP signalling ports (5060/2000).

CIPC uses a different cipher when doing the TLS handshake and requires the null-sha1 cipher and SSL encryption be configured. To add the null-shal cipher, use the show run all ssl command to see the output for the ssl encryption command and add null-shal to the end of the SSL encryption list.

NoteWhen used with CIPC, the phone proxy does not support end-users resetting their device name in CIPC (Preferences > Network tab > Use this Device Name field) or Administrators resetting the device name in Cisco Unified CM Administration console (Device menu > Phone Configuration > Device Name field). To function with the phone proxy, the CIPC configuration file must be in the format: SEP<mac_address>.cnf.xml. If the device name does not follow this format (SEP<mac_address>), CIPC cannot retrieve its configuration file from Cisco UMC via the phone proxy and CIPC will not function. When used with CIPC, the phone proxy does not support end-users resetting their device name in CIPC (Preferences > Network tab > Use this Device Name field) or Administrators resetting the device name in Cisco Unified CM Administration console (Device menu > Phone Configuration > Device Name field). To function with the phone proxy, the CIPC configuration file must be in the format: SEP<mac_address>.cnf.xml. If the device name does not follow this format (SEP<mac_address>), CIPC cannot retrieve its configuration file from Cisco UMC via the phone proxy and CIPC will not function.

Prerequisites for Rate Limiting TFTP Requests

In a remote access scenario, we recommend that you configure rate limiting of TFTP requests because any IP phone connecting through the Internet is allowed to send TFTP requests to the TFTP server.

To configure rate limiting of TFTP requests, configure the police command in the Modular Policy Framework. See the command reference for information about using the police command.

Policing is a way of ensuring that no traffic exceeds the maximum rate (in bits/second) that you configure, thus ensuring that no one traffic flow can take over the entire resource. When traffic exceeds the maximum rate, the ASA drops the excess traffic. Policing also sets the largest single burst of traffic allowed.

Rate Limiting Configuration Example

The following example describes how you configure rate limiting for TFTP requests by using the police command and the Modular Policy Framework.

Begin by determining the conformance rate that is required for the phone proxy. To determine the conformance rate, use the following formula:

X * Y * 8

Where

X = requests per second

Y = size of each packet, which includes the L2, L3, and L4 plus the payload

Therefore, if a rate of 300 TFTP requests/second is required, then the conformance rate would be calculated as follows:

300 requests/second * 80 bytes * 8 = 192000

The example configuration below shows how the calculated conformance rate is used with the police command:

access-list tftp extended permit udp any host 192.168.0.1 eq tftp

class-map tftpclass

match access-list tftp

policy-map tftpmap

class tftpclass

police output 192000

service-policy tftpmap interface inside

About ICMP Traffic Destined for the Media Termination Address

To control which hosts can ping the media termination address, use the icmp command and apply the access rule to the outside interface on the ASA.

Any rules for ICMP access applied to the outside interface apply to traffic destined for the media termination address.

For example, use the following command to deny ICMP pings from any host destined for the media termination address:

icmp deny any outside

To control which hosts can ping the media termination address, create an ICMP rule. Go to Configuration > Device Management > Management Access > ICMP and click the Add button.

End-User Phone Provisioning

The phone proxy is a transparent proxy with respect to the TFTP and signaling transactions. If NAT is not configured for the Cisco UCM TFTP server, then the IP phones need to be configured with the Cisco UCM cluster TFTP server address.

If NAT is configured for the Cisco UCM TFTP server, then the Cisco UCM TFTP server global address is configured as the TFTP server address on the IP phones.

Ways to Deploy IP Phones to End Users

In both options, deploying a remote IP phone behind a commercial Cable/DSL router with NAT capabilities is supported.

Option 1 (Recommended)

Stage the IP phones at corporate headquarters before sending them to the end users:

The phones register inside the network. IT ensures there are no issues with the phone configurations, image downloads, and registration.

If Cisco UCM cluster was in mixed mode, the CTL file should be erased before sending the phone to the end user.

Advantages of this option are:

Easier to troubleshoot and isolate problems with the network or phone proxy because you know whether the phone is registered and working with the Cisco UCM.

Better user experience because the phone does not have to download firmware from over a broadband connection, which can be slow and require the user to wait for a longer time.

Option 2

Send the IP phone to the end user. When using option 2, the user must be provided instructions to change the settings on phones with the appropriate Cisco UCM and TFTP server IP address.

NoteAs an alternative to authenticating remote IP phones through the TLS handshake, you can configure authentication via LSC provisioning. With LSC provisioning you create a password for each remote IP phone user and each user enters the password on the remote IP phones to retrieve the LSC. As an alternative to authenticating remote IP phones through the TLS handshake, you can configure authentication via LSC provisioning. With LSC provisioning you create a password for each remote IP phone user and each user enters the password on the remote IP phones to retrieve the LSC.

Because using LSC provisioning to authenticate remote IP phones requires the IP phones first register in nonsecure mode, Cisco recommends LSC provisioning be done inside the corporate network before giving the IP phones to end-users. Otherwise, having the IP phones register in nonsecure mode requires the Administrator to open the nonsecure signaling port for SIP and SCCP on the ASA.

Phone Proxy Guidelines and Limitations

General Guidelines and Limitations

The phone proxy has the following general limitations:

Only one phone proxy instance can be configured on the ASA by using the phone-proxy command. See the command reference for information about the phone-proxy command. See also Creating the Phone Proxy Instance.

The phone proxy only supports one Cisco UCM cluster. See Creating the CTL File for the steps to configure the Cisco UCM cluster for the phone proxy.

The phone proxy is not supported when the ASA is running in transparent mode or multiple context mode.

When a remote IP phone calls an invalid internal or external extension, the phone proxy does not support playing the annunciator message from the Cisco UCM. Instead, the remote IP phone plays a fast busy signal instead of the annunciator message "Your call cannot be completed ..." However, when an internal IP phone dials in invalid extension, the annunciator messages plays "Your call cannot be completed ..."

Packets from phones connecting to the phone proxy over a VPN tunnel are not inspected by the ASA inspection engines.

The phone proxy does not support IP phones sending Real-Time Control Protocol (RTCP) packets through the ASA. Disable RTCP packets in the Cisco Unified CM Administration console from the Phone Configuration page. See your Cisco Unified Communications Manager (CallManager) documentation for information about setting this configuration option.

When used with CIPC, the phone proxy does not support end-users resetting their device name in CIPC (Preferences > Network tab > Use this Device Name field) or Administrators resetting the device name in Cisco Unified CM Administration console (Device menu > Phone Configuration > Device Name field). To function with the phone proxy, the CIPC configuration file must be in the format: SEP<mac_address>.cnf.xml. If the device name does not follow this format (SEP<mac_address>), CIPC cannot retrieve its configuration file from Cisco UMC via the phone proxy and CIPC will not function.

The phone proxy does not support IP phones sending SCCP video messages using Cisco VT Advantage because SCCP video messages do not support SRTP keys.

For mixed-mode clusters, the phone proxy does not support the Cisco Unified Call Manager using TFTP to send encrypted configuration files to IP phones through the ASA.

Multiple IP phones behind one NAT device must be configured to use the same security mode.

When the phone proxy is configured for a mixed-mode cluster and multiple IP phones are behind one NAT device and registering through the phone proxy, all the SIP and SCCP IP phones must be configured as authenticated or encrypted, or all as non-secure on the Unified Call Manager.

For example, if there are four IP phones behind one NAT device where two IP phones are configured using SIP and two IP phones are configured using SCCP, the following configurations on the Unified Call Manager are acceptable:

– Two SIP IP phones: one IP phone in authenticated mode and one in encrypted mode, both in authenticated mode, or both in encrypted mode

Two SCCP IP phones: one IP phone in authenticated mode and one in encrypted mode, both in authenticated mode, or both in encrypted mode

– Two SIP IP phones: both in non-secure mode

Two SCCP IP phones: one IP phone in authenticated mode and one in encrypted mode, both in authenticated mode, both in encrypted mode

– Two SIP IP phones: one IP phone in authenticated mode and one in encrypted mode, both in authenticated mode, both in encrypted mode

Two SCCP IP phones: both in non-secure mode

This limitation results from the way the application-redirect rules (rules that convert TLS to TCP) are created for the IP phones.

Media Termination Address Guidelines and Limitations

The phone proxy has the following limitations relating to configuring the media-termination address:

When configuring the media-termination address, the phone proxy does not support having internal IP phones (IP phones on the inside network) being on a different network interface from the Cisco UCM unless the IP phones are forced to use the non-secure Security mode.

When internal IP phones are on a different network interface than the Cisco UCM, the IP phones signalling sessions still go through ASA; however, the IP phone traffic does not go through the phone proxy. Therefore, Cisco recommends that you deploy internal IP phones on the same network interface as the Cisco UMC.

If the Cisco UMC and the internal IP phones must be on different network interfaces, you must add routes for the internal IP phones to access the network interface of the media-termination address where Cisco UMC resides.

When the phone proxy is configured to use a global media-termination address, all IP phones see the same global address, which is a public routable address.

If you decide to configure a media-termination address on interfaces (rather than using a global interface), you must configure a media-termination address on at least two interfaces (the inside and an outside interface) before applying the phone-proxy service policy. Otherwise, you will receive an error message when enabling the Phone Proxy with SIP and Skinny Inspection.

The phone proxy can use only one type of media termination instance at a time; for example, you can configure a global media-termination address for all interfaces or configure a media-termination address for different interfaces. However, you cannot use a global media-termination address and media-termination addresses configured for each interface at the same time.

Task Flow for Configuring the Phone Proxy in a Non-secure Cisco UCM Cluster

Follow these tasks to configure the phone proxy in a Non-secure Cisco UCM Cluster:

Step 1 Create trustpoints and generate certificates for each entity in the network (Cisco UCM, Cisco UCM and TFTP, TFTP server, CAPF) that the IP phone must trust. The certificates are used in creating the CTL file. See Creating Trustpoints and Generating Certificates.

Importing Certificates from the Cisco UCM

For the TLS proxy used by the phone proxy to complete the TLS handshake successfully, it needs to verify the certificates from the IP phone (and the Cisco UCM if doing TLS with Cisco UCM). To validate the IP phone certificate, we need the CA Manufacturer certificate which is stored on the Cisco UCM. Follow these steps to import the CA Manufacturer certificate to the ASA.

Step 1Go to the Cisco UCM Operating System Administration web page.

Step 2 Choose Security > Certificate Management .

Note Earlier versions of Cisco UCM have a different UI and way to locate the certificates. For example, in Cisco UCM version 4.x, certificates are located in the directory C:\Program Files\Cisco\Certificates. See your Cisco Unified Communications Manager (CallManager) documentation for information about locating certificates.

Step 3 Click Find and it will display all the certificates.

Step 4Find the filename Cisco_Manufacturing_CA . This is the certificate need to verify the IP phone certificate. Click the .PEM file Cisco_Manufacturing_CA.pem . This will show you the certificate information and a dialog box that has the option to download the certificate.

Note If the certificate list contains more than one certificate with the filename Cisco_Manufacturing_CA, make you select the certificate Cisco_Manufacturing_CA.pem—the one with the .pem file extension.

Step 5 Click Download and save the file as a text file.

Step 6 On the ASA, create a trustpoint for the Cisco Manufacturing CA and enroll via terminal by entering the following commands. Enroll via terminal because you will paste the certificate you downloaded in Step 4.

hostname(config)# crypto ca trustpointtrustpoint_name

hostname(config-ca-trustpoint)# enrollment terminal

Step 7 Authenticate the trustpoint by entering the following command:

hostname(config)# crypto caauthenticatetrustpoint

Step 8You are prompted to “Enter the base 64 encoded CA Certificate.” Copy the .PEM file you downloaded in Step 4 and paste it at the command line. The file is already in base-64 encoding so no conversion is required. If the certificate is OK, you are prompted to accept it: “Do you accept this certificate? [yes/no].” Enter yes .

Note When you copy the certificate, make sure that you also copy also the lines with BEGIN and END.

Tip If the certificate is not ok, use the debug crypto ca command to show debug messages for PKI activity (used with CAs).

Step 9 Repeat the Step 1 through Step 8 for the next certificate. Table 3-2 shows the certificates that are required by the ASA.

Table 3-2 Certificates Required by the Security Appliance for the Phone Proxy

Certificate Name

Required for...

CallManager

Authenticating the Cisco UCM during TLS handshake; only required for mixed-mode clusters.

Task Flow for Configuring the Phone Proxy in a Mixed-mode Cisco UCM Cluster

NoteFor mixed-mode clusters, the phone proxy does not support the Cisco Unified Call Manager using TFTP to send encrypted configuration files to IP phones through the ASA. For mixed-mode clusters, the phone proxy does not support the Cisco Unified Call Manager using TFTP to send encrypted configuration files to IP phones through the ASA.

Follow these tasks to configure the phone proxy in a Non-secure Cisco UCM Cluster:

Step 1 Create trustpoints and generate certificates for each entity in the network (Cisco UCM, Cisco UCM and TFTP, TFTP server, CAPF) that the IP phone must trust. The certificates are used in creating the CTL file. See Creating Trustpoints and Generating Certificates.

Step 6 While configuring the phone proxy instance (in the Phone Proxy Configuration mode), enter the following command to configure the mode of the cluster to be mixed mode because the default is nonsecure:

Creating Trustpoints and Generating Certificates

Create trustpoints and generate certificates for each entity in the network (Cisco UCM, Cisco UCM and TFTP, TFTP server, CAPF) that the IP phone must trust. The certificates are used in creating the CTL file.

You need to create trustpoints for each Cisco UCM (primary and secondary if a secondary Cisco UCM is used) and TFTP server in the network. The trustpoints need to be in the CTL file for the phones to trust the Cisco UCM.

Creating the CTL File

Create the CTL file that will be presented to the IP phones during the TFTP requests.

Prerequisites

If you are using domain names for your Cisco UCM and TFTP server, you must configure DNS lookup on the ASA. Add an entry for each of the outside interfaces on the ASA into your DNS server, if such entries are not already present. Each ASA outside IP address should have a DNS entry associated with it for lookups. These DNS entries must also be enabled for Reverse Lookup.

Note You can enter the dns domain-lookup command multiple times to enable DNS lookup on multiple interfaces. If you enter multiple commands, the ASA tries each interface in the order it appears in the configuration until it receives a response.

See the command reference for information about the dns domain-lookup command.

Using an Existing CTL File

NoteOnly when the phone proxy is running in mixed-mode clusters, you have the option to use an existing CTL file to install trustpoints. Only when the phone proxy is running in mixed-mode clusters, you have the option to use an existing CTL file to install trustpoints.

If you have an existing CTL file that contains the correct IP addresses of the entities (namely, the IP address that the IP phones use for the Cisco UCM or TFTP servers), you can be use it to create a new CTL file thereby using the existing CTL file to install the trustpoints for each entity in the network (Cisco UCM, Cisco UCM and TFTP, TFTP server, CAPF) that the IP phones must trust.

Prerequisites

If a CTL file exists for the cluster, copy the CTL file to Flash memory. When you copy the CTL file to Flash memory, rename the file and do not name the file CTLFile.tlv .

If you are using domain names for your Cisco UCM and TFTP server, you must configure DNS lookup on the ASA. See the prerequisites for Creating the CTL File.

Command

Purpose

Step 1

hostname(config)#
ctl-file ctl_name

Example:

ctl-file myctl

Creates the CTL file instance.

Step 2

hostname(config-ctl-file)#
cluster-ctl-file filename_path

Example:

hostname(config-ctl-file)# cluster-ctl-file disk0:/old_ctlfile.tlv

Uses the trustpoints that are already in the existing CTL file stored in Flash memory.

Where the existing CTL file was saved to Flash memory with a filename other than CTLFile.tlv ; for example, old_ctlfile.tlv .

What to Do Next

When using an existing CTL file to configure the phone proxy, you can add additional entries to the file as necessary. See Creating the CTL File.

Exports the local CA certificate and installs it as a trusted certificate on the Cisco Unified Communications Manager server by performing one of the following actions.

hostname(config)#
crypto ca export trustpoint identity-certificate

Example:

hostname(config)# crypto ca export ldc_server identity-certificate

Exports the certificate if a trustpoint with proxy-ldc-issuer is used as the signer of the dynamic certificates.

hostname(config)#
show crypto ca server certificates

Exports the certificate for the embedded local CA server LOCAL-CA-SERVER.

After exporting the certificate, you must save the output to a file and import it on the Cisco Unified Communications Manager. You can use the Display Certificates function in the Cisco Unified Communications Manager software to verify the installed certificate.

For information about performing these procedures, see the following URLs:

Configures the media-termination address used by the media termination instance. The phone proxy uses this address for SRTP and RTP.

For the media termination instance, you can configure a global media-termination address for all interfaces or configure a media-termination address for different interfaces. However, you cannot use a global media-termination address and media-termination addresses configured for each interface at the same time.

If you configure a media termination address for multiple interfaces, you must configure an address on each interface that the ASA uses when communicating with IP phones.

The IP addresses are publicly routable addresses that are unused IP addresses within the address range on that interface.

(Optional) If the operational environment has an external HTTP proxy to which the IP phones direct all HTTP request, configures a proxy server.

You can configure only one proxy server while the phone proxy is in use.

By default, the Phone URL Parameters configured under the Enterprise Parameters use an FQDN in the URLs. The parameters might need to be changed to use an IP address if the DNS lookup for the HTTP proxy does not resolve the FQDNs.

Note If the IP phones have already downloaded their configuration files after you have configured the proxy server, you must restart the IP phones so that they get the configuration file with the proxy server address in the file.

NoteBefore you enable SIP and Skinny inspection for the Phone Proxy (which is done by applying the Phone Proxy to a service policy rule), the Phone Proxy must have an MTA instance, TLS Proxy, and CTL file assigned to it before the Phone Proxy can be applied to a service policy. Additionally, once a Phone Proxy is applied to a service policy rule, the Phone Proxy cannot be changed or removed. Before you enable SIP and Skinny inspection for the Phone Proxy (which is done by applying the Phone Proxy to a service policy rule), the Phone Proxy must have an MTA instance, TLS Proxy, and CTL file assigned to it before the Phone Proxy can be applied to a service policy. Additionally, once a Phone Proxy is applied to a service policy rule, the Phone Proxy cannot be changed or removed.

Step 5: Enable the Phone Proxy with SIP and Skinny inspection.

Creating the CTL File

Create a Certificate Trust List (CTL) file that is required by the Phone Proxy. Specify the certificates needed by creating a new CTL file or by specifying the path of an exiting CTL file to parse from Flash memory.

Create trustpoints and generate certificates for each entity in the network (CUCM, CUCM and TFTP, TFTP server, CAPF) that the IP phones must trust. The certificates are used in creating the CTL file. You need to create trustpoints for each CUCM (primary and secondary if a secondary CUCM is used) and TFTP server in the network. The trustpoints need to be in the CTL file for the phones to trust the CUCM.

Create the CTL File that will be presented to the IP phones during the TFTP. The address must be the translated or global address of the TFTP server or CUCM if NAT is configured.

When the file is created, it creates an internal trustpoint used by the Phone Proxy to sign the TFTP files. The trustpoint is named _internal_PP_ ctl-instance_filename .

NoteWhen a CTL file instance is assigned to the Phone Proxy, you cannot modify it in the CTL File pane and the pane is disabled. To modify a CTL File that is assigned to the Phone Proxy, go to the Phone Proxy pane (Configuration > Firewall > Unified Communications > Phone Proxy), and deselect the Use the Certificate Trust List File generated by the CTL instance check box. When a CTL file instance is assigned to the Phone Proxy, you cannot modify it in the CTL File pane and the pane is disabled. To modify a CTL File that is assigned to the Phone Proxy, go to the Phone Proxy pane (Configuration > Firewall > Unified Communications > Phone Proxy), and deselect the Use the Certificate Trust List File generated by the CTL instance check box.

Use the Create a Certificate Trust List (CTL) File pane to create a CTL file for the Phone Proxy. This pane creates the CTL file that is presented to the IP phones during the TFTP handshake with the ASA. For a detailed overview of the CTL file used by the Phone Proxy, see Creating the CTL File.

The Create a Certificate Trust List (CTL) File pane is used to configure the attributes for generating the CTL file. The name of the CTL file instance is generated by the ASDM. When the user tries to edit the CTL file instance configuration, the ASDM automatically generates the shutdown CLI command first and the no shutdown CLI command as the last command.

This pane is available from the Configuration > Firewall > Unified Communications > CTL File pane.

Step 3 To specify the CTL file to use for the Phone Proxy, perform one of the following:

If there is an existing CTL file available, download the CTL file to Flash memory by using the File Management Tool in the ASDM Tools menu. Select the Use certificates present in the CTL stored in flash radio button and specify the CTL file name and path in the text box.

Use an existing CTL file to install the trustpoints for each entity in the network (CUCM, CUCM and TFTP, TFTP server, CAPF) that the IP phones must trust. If you have an existing CTL file that contains the correct IP addresses of the entities (namely, the IP address that the IP phones use for the CUCM or TFTP servers), you can be use it to create a new CTL file. Store a copy of the existing CTL file to Flash memory and rename it something other than CTLFile.tlv

If there is no existing CTL file available, select Create new CTL file radio button.

Because the Phone Proxy generates the CTL file, it needs to create the System Administrator Security Token (SAST) key to sign the CTL file itself. This key can be generated on the ASA. A SAST is created as a self-signed certificate. Typically, a CTL file contains more than one SAST. In case a SAST is not recoverable, the other one can be used to sign the file later.

Step 5 Click Apply to save the CTL file configuration settings.

Adding or Editing a Record Entry in a CTL File

NoteThis feature is not supported for the Adaptive Security Appliance version 8.1.2. This feature is not supported for the Adaptive Security Appliance version 8.1.2.

Use the Add/Edit Record Entry dialog box to specify the trustpoints to be used for the creation of the CTL file.

NoteYou can edit an entry in the CTL file by using the Edit Record Entry dialog box; however, changing a setting in this dialog box does not change related settings for the phone proxy. For example, editing the IP address for the CUCM or TFTP servers in this dialog changes the setting only in the CTL file and does not change the actual addresses of those servers or update the address translations required by the phone proxy. You can edit an entry in the CTL file by using the Edit Record Entry dialog box; however, changing a setting in this dialog box does not change related settings for the phone proxy. For example, editing the IP address for the CUCM or TFTP servers in this dialog changes the setting only in the CTL file and does not change the actual addresses of those servers or update the address translations required by the phone proxy. To modify CTL file settings, we strongly recommend you re-run the Unified Communications Wizard to edit CTL file settings and ensure proper synchronization with all phone proxy settings.

Add additional record-entry configurations for each entity that is required in the CTL file.

cucm: Specifies the role of this trustpoint to be CCM. Multiple CCM trustpoints can be configured.

cucm-tftp: Specifies the role of this trustpoint to be CCM+TFTP. Multiple CCM+TFTP trustpoints can be configured.

tftp: Specifies the role of this trustpoint to be TFTP. Multiple TFTP trustpoints can be configured.

capf: Specifies the role of this trustpoint to be CAPF. Only one CAPF trustpoint can be configured.

Step 4 In the Host field, specify the IP address of the trustpoint. The IP address you specify must be the global address of the TFTP server or CUCM if NAT is configured. The global IP address is the IP address as seen by the IP phones because it will be the IP address used for the CTL record for the trustpoint.

Step 5 In the Certificate field, specify the Identity Certificate for the record entry in the CTL file. You can create a new Identity Certificate by clicking Manage . The Manage Identify Certificates dialog box opens. See the general operations configuration guide.

You can add an Identity Certificate by generating a self-signed certificate, obtaining the certificate through SCEP enrollment, or by importing a certificate in PKCS-12 format. Choose the best option based on the requirements for configuring the CTL file.

Step 6 (Optional) In the Domain Name field, specify the domain name of the trustpoint used to create the DNS field for the trustpoint. This is appended to the Common Name field of the Subject DN to create the DNS Name. The domain name should be configured when the FQDN is not configured for the trustpoint. Only one domain-name can be specified.

NoteIf you are using domain names for your CUCM and TFTP server, you must configure DNS lookup on the ASA. Add an entry for each of the outside interfaces on the ASA into your DNS server, if such entries are not already present. Each ASA outside IP address should have a DNS entry associated with it for lookups. These DNS entries must also be enabled for Reverse Lookup. Additionally, define your DNS server IP address on the ASA; for example: If you are using domain names for your CUCM and TFTP server, you must configure DNS lookup on the ASA. Add an entry for each of the outside interfaces on the ASA into your DNS server, if such entries are not already present. Each ASA outside IP address should have a DNS entry associated with it for lookups. These DNS entries must also be enabled for Reverse Lookup. Additionally, define your DNS server IP address on the ASA; for example: dns name-server 10.2.3.4 (IP address of your DNS server).

Creating the Media Termination Instance

NoteIn versions before 8.2(1), you configured one media-termination address (MTA) on the outside interface of the adaptive security appliance where the remote Cisco IP phones were located. In Version 8.2(1) and later, you can configure a global media-termination address for all interfaces or configure a media-termination address for different interfaces. In versions before 8.2(1), you configured one media-termination address (MTA) on the outside interface of the adaptive security appliance where the remote Cisco IP phones were located. In Version 8.2(1) and later, you can configure a global media-termination address for all interfaces or configure a media-termination address for different interfaces.

As a result of this enhancement, the old configuration has been deprecated. You can continue to use the old configuration if desired. However, if you need to change the configuration at all, only the new configuration method is accepted; you cannot later restore the old configuration. If you need to maintain downgrade compatibility, you should keep the old configuration as is.

Step 3 In the Media Termination Address Settings area, specify whether to configure a media-termination address (MTA) per interface or to configure a global MTA. You can configure a global media-termination address for all interfaces or configure a media-termination address for different interfaces.

To configure an MTA per interface, click the Configure MTA per Interface radio button and click the Add button. In the dialog box that appears, specify the interface name and enter an IP address or hostname.

If you configure a media termination address for multiple interfaces, you must configure an address on each interface that the ASA uses when communicating with IP phones. The IP addresses are publicly routable addresses that are unused IP addresses within the address range on that interface.

To configure a global MTA, click the Configure global MTA on interface radio button and enter the IP address in the text box. See Media Termination Instance Prerequisites for the complete list of requirements that you must follow when configuring a global media termination address.

Step 4 Specify the minimum and maximum values for the RTP port range for the media termination instance. The minimum port and the maximum port can be a value from 1024 to 65535.

Creating the Phone Proxy Instance

Create the phone proxy instance. To have a fully functional phone proxy, you must also complete additional tasks, such as creating the MTA and enabling SIP and SCCP (Skinny) inspection. See Importing Certificates from the Cisco UCM for the complete list of tasks.

Prerequisites

You must have already created the CTL file and TLS proxy instance for the phone proxy.

Step 3 Check the Apply MTA instance to Phone Proxy check box to add the media termination address to the Phone Proxy instance. You must have a media termination address instance configured. The configured address is added to the Phone Proxy instance.

NoteThe TFTP server must reside on the same interface as the Cisco Unified Call Manager. Additionally, If NAT is configured for the TFTP server, the NAT configuration must be configured prior to configuring the specifying the TFTP server while creating the Phone Proxy instance. The TFTP server must reside on the same interface as the Cisco Unified Call Manager. Additionally, If NAT is configured for the TFTP server, the NAT configuration must be configured prior to configuring the specifying the TFTP server while creating the Phone Proxy instance.

Step 5 Specify the CTL File to use for the Phone Proxy by doing one of the following:

To use an existing CTL File, check the Use the Certificate Trust List File generated by the CTL instance check box.

To create a new CTL file for the Phone Proxy, click the link Generate Certificate Trust List File. The Create a Certificate Trust List (CTL) File pane opens. See Creating the CTL File.

Step 6 To specify the security mode of the CUCM cluster, click one of the following options in the CUCM Cluster Mode field:

Non-secure—Specifies the cluster mode to be in nonsecure mode when configuring the Phone Proxy feature.

Mixed—Specifies the cluster mode to be in mixed mode when configuring the Phone Proxy feature.

Step 7 To configure the idle timeout after which the secure-phone entry is removed from the Phone Proxy database (the default is 5 minutes), enter a value in the format hh : mm : ss .

Since secure phones always request a CTL file upon bootup, the Phone Proxy creates a database that marks the phone as secure. The entries in the secure phone database are removed after a specified configured timeout. The entry timestamp is updated for each registration refresh the Phone Proxy receives for SIP phones and KeepAlives for SCCP phones.

Specify a value that is greater than the maximum timeout value for SCCP KeepAlives and SIP Register refresh. For example, if the SCCP KeepAlives are configured for 1 minute intervals and the SIP Register Refresh is configured for 3 minutes, configure this timeout value greater than 3 minutes.

Step 8 To preserve Call Manager configuration on the IP phones, check the Preserve the Call Manager’s configuration on the phone... check box. When this check box is uncheck, the following service settings are disabled on the IP phones:

Because CIPC requires an LSC to perform the TLS handshake, CIPC needs to register with the CUCM in nonsecure mode using cleartext signaling. To allow the CIPC to register, create an ACL that allows the CIPC to connect to the CUCM on the nonsecure SIP/SCCP signalling ports (5060/2000).

CIPC uses a different cipher when doing the TLS handshake and requires the null-sha1 cipher and SSL encryption be configured. To add the null-shal cipher, go to Configuration > Device Management > Advanced > SSL Settings > Encryption section. Select the null-shal SSL encryption type and add it to the Available Algorithms.

Current versions of Cisco IP Communicator (CIPC) support authenticated mode and perform TLS signaling but not voice encryption.

Step 10 To configure an HTTP proxy for the Phone Proxy feature that is written into the IP phone's configuration file under the <proxyServerURL> tag, do the following:

a. Check the Configure a http-proxy which would be written into the phone’s config file... check box.

b. In the IP Address field, type the IP address of the HTTP proxy and the listening port of the HTTP proxy.

The IP address you enter should be the global IP address based on where the IP phone and HTTP proxy server is located. You can enter a hostname in the IP Address field when that hostname can be resolved to an IP address by the ASA (for example, DNS lookup is configured) because the ASA will resolve the hostname to an IP address. If a port is not specified, the default will be 8080.

c. In the Interface field, select the interface on which the HTTP proxy resides on the ASA.

Setting the proxy server configuration option for the Phone Proxy allows for an HTTP proxy on the DMZ or external network in which all the IP phone URLs are directed to the proxy server for services on the phones. This setting accommodates nonsecure HTTP traffic, which is not allowed back into the corporate network.

Step 11 Click Apply to save the Phone Proxy configuration settings.

NoteAfter creating the Phone Proxy instance, you enable it with SIP and Skinny inspection. After creating the Phone Proxy instance, you enable it with SIP and Skinny inspection.

However, before you enable SIP and Skinny inspection for the Phone Proxy (which is done by applying the Phone Proxy to a service policy rule), the Phone Proxy must have an MTA instance, TLS Proxy, and CTL file assigned to it before the Phone Proxy can be applied to a service policy. Additionally, once a Phone Proxy is applied to a service policy rule, the Phone Proxy cannot be changed or removed.

Adding or Editing the TFTP Server for a Phone Proxy

NoteThis feature is not supported for the Adaptive Security Appliance version 8.1.2. This feature is not supported for the Adaptive Security Appliance version 8.1.2.

NoteYou can edit the TFTP server setting by using the Edit TFTP Server dialog box; however, changing a setting in this dialog box does not change related settings for the phone proxy. For example, editing the IP address for the TFTP server in this dialog does not change the setting in the CTL file and does not update the address translations required by the phone proxy. You can edit the TFTP server setting by using the Edit TFTP Server dialog box; however, changing a setting in this dialog box does not change related settings for the phone proxy. For example, editing the IP address for the TFTP server in this dialog does not change the setting in the CTL file and does not update the address translations required by the phone proxy. To modify TFTP server settings, we strongly recommend you re-run the Unified Communications Wizard to ensure proper synchronization with all phone proxy settings.

Use the Add/Edit TFTP Server dialog box to specify the IP address of the TFTP server and the interface on which the TFTP server resides.

The Phone Proxy must have at least one CUCM TFTP server configured. Up to five TFTP servers can be configured for the Phone Proxy.

The TFTP server is assumed to be behind the firewall on the trusted network; therefore, the Phone Proxy intercepts the requests between the IP phones and TFTP server.

NoteIf NAT is configured for the TFTP server, the NAT configuration must be configured prior to specifying the TFTP server while creating the Phone Proxy instance. If NAT is configured for the TFTP server, the NAT configuration must be configured prior to specifying the TFTP server while creating the Phone Proxy instance.

Step 4 In the TFTP Server IP Address field, specify the address of the TFTP server. Create the TFTP server using the actual internal IP address.

Step 5 (Optional) In the Port field, specify the port the TFTP server is listening in on for the TFTP requests. This should be configured if it is not the default TFTP port 69.

Step 6 In the Interface field, specify the interface on which the TFTP server resides. The TFTP server must reside on the same interface as the Cisco Unified Call Manager (CUCM).

When IP phones are behind a NAT-capable router, the router can be configured to forward the UDP ports to the IP address of the IP phone. Specifically, configure the router for UDP port forwarding when an IP phone is failing during TFTP requests and the failure is due to the router dropping incoming TFTP data packets. Configure the router to enable UDP port forwarding on port 69 to the IP phone.

As an alternative of explicit UDP forwarding, some Cable/DSL routers require you to designate the IP phone as a DMZ host. For Cable/DSL routers, this host is a special host that receives all incoming connections from the public network.

When configuring the phone proxy, there is no functional difference between an IP phone that has UDP ports explicitly forwarded or an IP phone designated as a DMZ host. The choice is entirely dependent upon the capabilities and preference of the end user.

Configuring Your Router

Your firewall/router needs to be configured to forward a range of UDP ports to the IP phone. This will allow the IP phone to receive audio when you make/receive calls.

NoteDifferent Cable/DSL routers have different procedures for this configuration. Furthermore most NAT-capable routers will only allow a given port range to be forwarded to a single IP address Different Cable/DSL routers have different procedures for this configuration. Furthermore most NAT-capable routers will only allow a given port range to be forwarded to a single IP address

The configuration of each brand/model of firewall/router is different, but the task is the same. For specific instructions for your brand and model of router, please contact the manufacturer’s website.

Linksys Routers

Step 1 From your web browser, connect to the router administrative web page. For Linksys, this is typically something like http://192.168.1.1 .

Debugging Information from the Security Appliance

This section describes how to use the debug , capture , and show commands to obtain debugging information for the phone proxy. See the command reference for detailed information about the syntax for these commands.

Table 3-4 Security Appliance Debug Commands to Use with the Phone Proxy

To

Use the Command

Notes

To show error and event messages for TLS proxy inspection.

debug inspect tls-proxy [ events | errors ]

Use this command when your IP phone has successfully downloaded all TFTP files but is failing to complete the TLS handshake with the TLS proxy configured for the phone proxy.

To show error and event messages of media sessions for SIP and Skinny inspections related to the phone proxy.

debug phone-proxy media [ events | errors ]

Use this command in conjunction with the debug sip command and the debug skinny command if your IP phone is experiencing call failures or audio problems.

To show error and event messages of signaling sessions for SIP and Skinny inspections related to the phone proxy.

debug phone-proxy signaling [ events | errors ]

Use this command in conjunction with the debug sip command and the debug skinny command if your IP phone is failing to register with the Cisco UCM or if you are experiencing call failure.

To show error and event messages of TFTP inspection, including creation of the CTL file and configuration file parsing.

debug phone-proxy tftp [ events | errors ]

To show debug messages for SIP application inspection.

debug sip

Use this command when your IP phones are experiencing connection problems; for example, you can connect within the network but cannot make calls off the network. In the output, check for 4XX or 5XX messages.

To show debug messages for SCCP (Skinny) application inspection.

debug skinny

Use this command when your IP phones are experiencing connection problems; for example, you can connect within the network but cannot make calls off the network. In the output, check for 4XX or 5XX messages.

Table 3-5 lists the capture commands to use with the phone proxy. Use the capture command on the appropriate interfaces (IP phones and Cisco UCM) to enable packet capture capabilities for packet sniffing and network fault isolation.

Table 3-5 Security Appliance Capture Commands to Use with the Phone Proxy

To

Use the Command

Notes

To capture packets on the ASA interfaces.

capture capture_name interface interface_name

Use this command if you are experiencing any problems that might require looking into the packets.

For example, if there is a TFTP failure and the output from the debug command does not indicate the problem clearly, run the capture command on the interface on which the IP phone resides and the interface on which the TFTP server resides to see the transaction and where the problem could be.

To capture data from the TLS proxy when there is a non-secure IP phone connecting to the phone proxy on the inside interface.

Table 3-6 Security Appliance Show Commands to Use with the Phone Proxy

To

Use the Command

Notes

To show the packets or connections dropped by the accelerated security path.

show asp drop

Use this command to troubleshoot audio quality issues with the IP phones or other traffic issues with the phone proxy. In addition to running this command, get call status from the phone to check for any dropped packets or jitter. See Debugging Information from IP Phones.

To show the classifier contents of the accelerated security path for the specific classifier domain.

show asp table classify domain domain_name

If the IP phones are not downloading TFTP files, use this command to check that the classification rule for the domain inspect-phone-proxy is set for hosts to the configured TFTP server under the phone proxy instance.

If the IP phones are failing to register, use this command to make sure there is a classification rule for the domain app-redirect set for the IP phones that cannot register.

To show the connections that are to the ASA or from the ASA, in addition to through-traffic connections.

show conn all

If you are experiencing problems with audio, use this command to make sure that there are connections opened from the IP phone to the media termination address.

Using this command shows that the phone proxy has connections that are going through “inspect-phone-proxy”, which inspects TFTP connections. Using this command verifies that the TFTP requests are being inspected because the p flag is there.

To show the logs in the buffer and logging settings.

show logging

Before entering the show logging command, enable the logging buffered command so that the show logging command displays the current message buffer and the current settings.

Use this command to determine if the phone proxy and IP phones are successfully completing the TLS handshake.

Note Using the show logging command is useful for troubleshooting many problems where packets might be denied or there are translation failures.

To show the corresponding media sessions stored by the phone proxy.

show phone-proxy media-sessions

Use this command to display output from successful calls. Additionally, use this command to troubleshoot problems with IP phone audio, such as one-way audio.

To show the IP phones capable of Secure mode stored in the database.

show phone-proxy secure-phones

For any problems, make sure there is an entry for the IP phone in this output and that the port for this IP phone is non-zero, which indicates that it has successfully registered with the Cisco UCM.

To show the corresponding signaling sessions stored by the phone proxy.

show phone-proxy signaling-sessions

Use this command to troubleshoot media or signaling failure.

To show the configured service policies.

show service-policy

Use this command to show statistics for the service policy.

To show active TLS proxy sessions related to the phone proxy.

show tls-proxy sessions

If the IP phone has failed to register, use this command to see if the IP phone has successfully completed the handshake with the TLS proxy configured for the phone proxy.

Debugging Information from IP Phones

On the IP phone, perform the following actions:

Check the Status messages on the IP phone by selecting the Settings button > Status > Status Messages and selecting the status item that you want to view.

Collect the call-statistics data from the IP phone by selecting the Settings button > Status > Call Statistic. Data like the following displays:

RxType: G.729 TxType: G.729

RxSize: 20 ms TxSize: 20 ms

RxCnt: 0 TxCnt: 014174

AvgJtr: 10 MaxJtr: 59

RxDisc: 0000 RxLost: 014001

Check the Security settings on the IP phone by selecting the Settings button > Security Configuration. Settings for web access, Security mode, MIC, LSC, CTL file, trust list, and CAPF appear. Under Security mode, make sure the IP phone is set to Encrypted.

Check the IP phone to determine which certificates are installed on the phone by selecting the Settings button > Security Configuration > Trust List. In the trustlist, verify the following:

– Make sure that there is an entry for each entity that the IP phone will need to contact. If there is a primary and backup Cisco UCM, the trustlist should contain entries for each Cisco UCM.

– If the IP phone needs an LSC, the record entry should contain a CAPF entry.

– Make sure that the IP addresses listed for each entry are the mapped IP addresses of the entities that the IP phone can reach.

Open a web browser and access the IP phone console logs at the URL http:// IP_phone_IP address . The device information appears in the page. In the Device Logs section in the left pane, click Console Logs.

IP Phone Registration Failure

The following errors can make IP phones unable to register with the phone proxy:

TFTP Auth Error Displays on IP Phone Console

Solution This Status message can indicate a problem with the IP phone CTL file.

To correct problems with the IP phone CTL file, perform the following:

Step 1 From the IP phone, select the Setting button > Security Configuration > Trust List. Verify that each entity in the network—Primary Cisco UCM, Secondary Cisco UCM, TFTP server—has its own entry in the trustlist and that each entity IP address is reachable by the IP phone.

Step 2 From the ASA, verify that the CTL file for the phone proxy contains one record entry for each entity in the network—Primary Cisco UCM, Secondary Cisco UCM, TFTP server—by entering the following command:

ciscoasa# show running-config all ctl-file [ctl_name]

Each of these record entries creates one entry on the IP phone trustlist. The phone proxy creates one entry internally with the function CUCM+TFTP.

Solution The phone proxy should parse only the IP phone configuration file. When the phone proxy TFTP state gets out of state, the phone proxy cannot detect when it is attempting to parse a file other than the IP phone configuration file and the error above appears in the ASA output from the debug phone-proxy tftp command.

Perform the following actions to troubleshoot this problem:

Step 1 Reboot the IP phone.

Step 2 On the ASA, enter the following command to obtain the error information from the first TFTP request to the point where the first error occurred.

Step 4 Save this troubleshooting data, open a case with TAC and give them this information.

Cisco UCM Does Not Respond to TFTP Request for Configuration File

Problem When the ASA forwards the TFTP request to the Cisco UCM for the IP phone configuration file, the Cisco UCM does not respond and the following errors appear in the debug output ( debug phone-proxy tftp ):

IP Phone Requesting Unsigned File Error

Problem The IP phone should always request a signed file. Therefore, the TFTP file being requested always has the .SGN extension.

When the IP phone does not request a signed file, the following error appears in the debug output ( debug phone-proxy tftp errors ):

Error: phone requesting for unsigned config file

Solution Most likely, this error occurs because the IP phone has not successfully installed the CTL file from the ASA.

Determine whether the IP phone has successfully downloaded and installed the CTL file from the ASA by checking the Status messages on the IP phone. See Debugging Information from IP Phones for information.

IP Phone Unable to Download CTL File

Solution If the IP phone did not have an existing CTL file, check the Status messages by selecting the Settings button > Status > Status Messages. If the list contains a Status message indicating the IP phone encountered a CTL File Auth error, obtain the IP phone console logs, open a TAC case, and send them the logs.

Solution This error can appear in the IP phone Status messages when the IP phone already has an existing CTL file.

Step 1 Check the IP phone to see if a CTL file already exists on it. This can occur if the IP phone previously registered with a mixed mode cluster Cisco UCM. On the IP phone, select the Settings button > Security Configuration > CTL file.

Solution Problems downloading the CTL file might be caused by issues with media termination. Enter the following command to determine if the media-termination address in the phone proxy configuration is set correctly:

hostname(config)# show running-config all phone-proxy

!

phone-proxy mypp

media-termination address 10.10.0.25

cipc security-mode authenticated

cluster-mode mixed

disable service-settings

timeout secure-phones 0:05:00

hostname(config)#

Make sure that each media-termination instance is created correctly and that the address or addresses are set correctly. The ASA must meet specific criteria for media termination. See Media Termination Instance Prerequisites for the complete list of prerequisites that you must follow when creating the media termination instance and configuring the media termination addresses.

IP Phone Registration Failure from Signaling Connections

Problem The IP phone is unable to complete the TLS handshake with the phone proxy and download its files using TFTP.

Solution

Step 1 Determine if the TLS handshake is occurring between the phone proxy and the IP phone, perform the following:

a. Enable logging with the following command:

hostname(config)# logging buffered debugging

b. To check the output from the syslogs captured by the logging buffered command, enter the following command:

hostname# show logging

The syslogs will contain information showing when the IP phone is attempting the TLS handshake, which happens after the IP phone downloads its configuration file.

Step 2 Determine if the TLS proxy is configured correctly for the phone proxy:

a.Display all currently running TLS proxy configurations by entering the following command:

hostname# show running-config tls-proxy

tls-proxy proxy

server trust-point _internal_PP_<ctl_file_instance_name>

client ldc issuer ldc_signer

client ldc key-pair phone_common

no client cipher-suite

hostname#

b. Verify that the output contains the server trust-point command under the tls-proxy command (as shown in substep a.).

If you are missing the server trust-point command, modify the TLS proxy in the phone proxy configuration.

Having this command missing from the TLS proxy configuration for the phone proxy will cause TLS handshake failure.

Step 3 Verify that all required certificates are imported into the ASA so that the TLS handshake will succeed.

a. Determine which certificates are installed on the ASA by entering the following command:

hostname# show running-config crypto

Additionally, determine which certificates are installed on the IP phones. See Debugging Information from IP Phones for information about checking the IP phone to determine if it has MIC installed on it.

b. Verify that the list of installed certificates contains all required certificates for the phone proxy.

Step 4 If the steps above fail to resolve the issue, perform the following actions to obtain additional troubleshooting information for Cisco Support.

a. Enter the following commands to capture additional debugging information for the phone proxy:

hostname# debug inspect tls-proxy error

hostname# show running-config ssl

hostname(config) show tls-proxytls_namesessionhost host_addrdetail

b. Enable the capture command on the inside and outside interfaces (IP phones and Cisco UCM) to enable packet capture capabilities for packet sniffing and network fault isolation. See the command reference for information.

Problem The TLS handshake succeeds, but signaling connections are failing.

Solution Perform the following actions:

Check to see if SIP and Skinny signaling is successful by using the following commands:

– debug sip

– debug skinny

If the TLS handshake is failing and you receive the following syslog, the SSL encryption method might not be set correctly:

%ASA-3-717027: Certificate chain failed validation. No suitable trustpoint was found to validate chain.

Solution

Verify that all required certificates are imported into the ASA so that the TLS handshake will succeed.

Step 1 Determine which certificates are installed on the ASA by entering the following command:

hostname# show running-config crypto

Additionally, determine which certificates are installed on the IP phones. See Debugging Information from IP Phones for information about checking the IP phone to determine if it has MIC installed on it.

Step 2 Verify that the list of installed certificates contains all required certificates for the phone proxy.

In order for the phone proxy to authenticate the MIC provided by the IP phone, it needs the Cisco Manufacturing CA (MIC) certificate imported into the ASA.

Verify that all required certificates are imported into the ASA so that the TLS handshake will succeed.

Step 1 Determine which certificates are installed on the ASA by entering the following command:

hostname# show running-config crypto

Additionally, determine which certificates are installed on the IP phones. The certificate information is shown under the Security Configuration menu. See Debugging Information from IP Phones for information about checking the IP phone to determine if it has the MIC installed on it.

Step 2 Verify that the list of installed certificates contains all required certificates for the phone proxy.

Media Termination Address Errors

Problem Entering the media-termination address command displays the following errors:

hostname(config-phone-proxy)# media-termination addressip_address

ERROR: Failed to apply IP address to interface Virtual254, as the network overlaps with interface GigabitEthernet0/0. Two interfaces cannot be in the same subnet.

ERROR: Failed to set IP address for the Virtual interface

ERROR: Could not bring up Phone proxy media termination interface

ERROR: Failed to find the HWIDB for the Virtual interface

Solution Enter the following command to determine if the media-termination address in the phone proxy configuration is set correctly:

hostname(config)# show running-config all phone-proxy

asa2(config)# show running-config all phone-proxy

!

phone-proxy mypp

media-termination address 10.10.0.25

cipc security-mode authenticated

cluster-mode mixed

disable service-settings

timeout secure-phones 0:05:00

hostname(config)#

Make sure that each media-termination instance is created correctly and that the address or addresses are set correctly. The ASA must meet specific criteria for media termination. See Media Termination Instance Prerequisites for the complete list of prerequisites that you must follow when creating the media termination instance and configuring the media termination addresses.

Audio Problems with IP Phones

The following audio errors can occur when the IP phones connecting through the phone proxy.

Media Failure for a Voice Call

Problem The call signaling completes but there is one way audio or no audio.

Solution

Problems with one way or no audio might be caused by issues with media termination. Enter the following command to determine if the media-termination address in the phone proxy configuration is set correctly:

hostname(config)# show running-config all phone-proxy

asa2(config)# show running-config all phone-proxy

!

phone-proxy mypp

media-termination address 10.10.0.25

cipc security-mode authenticated

cluster-mode mixed

disable service-settings

timeout secure-phones 0:05:00

hostname(config)#

Make sure that each media-termination instance is created correctly and that the address or addresses are set correctly. The ASA must meet specific criteria for media termination. See Media Termination Instance Prerequisites for the complete list of prerequisites that you must follow when creating the media termination instance and configuring the media termination addresses.

If each media-termination address meets the requirements, determine whether the IP addresses are reachable by all IP phones.

If each IP address is set correctly and reachable by all IP phones, check the call statistics on an IP phone (see Debugging Information from IP Phones) and determine if there are Rcvr packets and Sender packets on the IP phone, or if there are any Rcvr Lost or Discarded packets.

Saving SAST Keys

Site Administrator Security Token (SAST) keys on the ASA can be saved in the event a recovery is required due to hardware failure and a replacement is required. The following steps shows how to recover the SAST keys and use them on the new hardware.

The SAST keys can be seen via the show crypto key mypubkey rsa command. The SAST keys are associated with a trustpoint that is labeled _internal_ ctl-file_name _SAST_ X where ctl-file-name is the name of the CTL file instance that was configured, and X is an integer from 0 to N-1 where N is the number of SASTs configured for the CTL file (the default is 2).

Step 1On the ASA, export all the SAST keys in PKCS-12 format by using the crypto ca export command:

Where trustpoint is _internal_ ctl-file_name _SAST_ X and ctl-file-name is the name of the CTL file instance that was configured, and X is an integer from 0 to 4 depending on what you exported from the ASA.

b. Using the PKCS-12 output you saved in Step 1, enter the following command and paste the output when prompted:

Step 3 Create the CTL file instance on the new ASA using the same name as the one used in the SAST trustpoints created in Step 2 by entering the following commands. Create trustpoints for each Cisco UMC (primary and secondary).

Figure 3-5 shows an example of the configuration for a mixed-mode Cisco UCM cluster using the following topology where the TFTP server resides on a different server from the primary and secondary Cisco UCMs.

In this sample, the static interface PAT for the TFTP server is configured to appear like the ASA’s outside interface IP address.

Figure 3-6 shows an example of the configuration for a mixed-mode Cisco UCM cluster where LSC provisioning is required using the following topology.

NoteDoing LSC provisioning for remote IP phones is not recommended because it requires that the IP phones first register and they have to register in nonsecure mode. Having the IP phones register in nonsecure mode requires the Administrator to open the nonsecure signaling port for SIP and SCCP on the ASA. If possible, LSC provisioning should be done inside the corporate network before giving the IP phones to the end-users. Doing LSC provisioning for remote IP phones is not recommended because it requires that the IP phones first register and they have to register in nonsecure mode. Having the IP phones register in nonsecure mode requires the Administrator to open the nonsecure signaling port for SIP and SCCP on the ASA. If possible, LSC provisioning should be done inside the corporate network before giving the IP phones to the end-users.

In this sample, you create an ACL to allow the IP phones to contact the TFTP server and to allow the IP phones to register in nonsecure mode by opening the nonsecure port for SIP and SCCP as well as the CAPF port for LSC provisioning.

Additionally, you create the CAPF trustpoint by copying and pasting the CAPF certificate from the Cisco UCM Certificate Management software.

Example 6: VLAN Transversal

Figure 3-7 shows an example of the configuration to force Cisco IP Communicator (CIPC) softphones to operate in authenticated mode when CIPC softphones are deployed in a voice and data VLAN scenario. VLAN transversal is required between CIPC softphones on the data VLAN and hard phones on the voice VLAN.

In this sample, the Cisco UCM cluster mode is nonsecure.

In this sample, you create an ACL to allow the IP phones to contact the TFTP server and to allow the IP phones to register in nonsecure mode by opening the nonsecure port for SIP and SCCP as well as the CAPF port for LSC provisioning.

In this sample, you configure NAT for the CIPC by using PAT so that each CIPC is mapped to an IP address space in the Voice VLAN.

Additionally, you create the CAPF trustpoint by copying and pasting the CAPF certificate from the Cisco UCM Certificate Management software.

NoteCisco IP Communicator supports authenticated mode only and does not support encrypted mode; therefore, there is no encrypted voice traffic (SRTP) flowing from the CIPC softphones. Cisco IP Communicator supports authenticated mode only and does not support encrypted mode; therefore, there is no encrypted voice traffic (SRTP) flowing from the CIPC softphones.

Figure 3-7 VLAN Transversal Between CIPC Softphones on the Data VLAN and Hard Phones on the Voice VLAN